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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Michal Piotrowski 29 Apr '07

29 Apr '07

Hi, On 29/04/07, Giridhar Pemmasani <pgiri(a)yahoo.com> wrote: > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails to > boot - it seems to stop at various points depending on what other modules are > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with everything > else being same, using maestro3.c from 2.6.20 kernel and using it in 2.6.21 > kernel also works - i.e., system boots fine. I will try to isolate the issue > and follow up. Please revert this patches http://www.stardust.webpages.pl/files/patches/alsa_revert/ and test kernel after each revert. Takashi, I'll add this to known regressions for -stable team. > > Giri Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 29 Apr '07

29 Apr '07

hi, I have an Acer Aspire 5100. I can play sound but with the build in mic, I can't record sound. The driver is the latest HG code. http://pastebin.ca/464406 [root@localhost ~]# lspci -nnv -d1002:437b 00:14.2 0403: 1002:437b (rev 01) Subsystem: 1025:009f Flags: bus master, slow devsel, latency 64, IRQ 19 Memory at c0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 [root@localhost ~]# vi /tmp/1 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/29/2007 04:18 AM, Cody Jung wrote: > Er...that appears to be a .diff file. Do I have to do something with it > for it to be workable, or...? Yes. You'd have to apply the patch (as it's called) to a recent kernel source tree, reconfigure the kernel adding this new VIVO driver to your config and then recompile and reinstall that kernel. Oh well, you posting to alsa-user and not alsa-devel should've been a hint. Rest assured the driver would've worked nicely :-/ There is another thing you can do. What was posted was just a wrapper around cs4231-lib in much the same way that the existing snd-cs4231 module is and that module you _will_ already have installed. That one doesn't do anything automatic though, so you will have to know how to manually enable your card first. First, open a root shell and look through the /sys/bus/pnp/devices directory for your VIVO. Under the "xx:yy.zz" sub-directories there you will find "id" files, and one of them will say "ENS1011". Example: # cat /sys/bus/pnp/devices/01\:01.00/id ENS1011 After having found it in there, activate the card: # echo activate >/sys/bus/pnp/devices/01\:01.00/resources and look at what resources were assigned to it by the PnP layer: # cat /sys/bus/pnp/devices/01\:01.00/resources state = active io 0x330-0x33f io 0x530-0x537 io 0x2100-0x213f irq 5 irq 9 dma 1 dma 0 You can now load the snd-cs4231 module specifying those card resources manually: # modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 Yes, 0x534. The second io resource + 4. This should've loaded without problem. Start up alsamixer: # alsamixer and up and unmute the PCM volume. With -> and <- you can walk through the sliders; ^ ups the one you're on, and "m" toggles the mute (the "MM" at the bottom should turn to a highlighted "OO"). PCM is the first slider. 90 seems to be an okay value. Quit alsamixer again with Alt-Q and now you should be able to hear things. If you have a plain "wav" file around somewhere, that makes for the quickest test: # aplay foo.wav No, I'm not going to tell you to plug in your speakers. If anything in the above doesn't work as expected, your system is not as expected and there's isn't much I can help with. If everything does work, you can automate it for the next time. I've attached (to try and avoid linewrap) the lines that you will need in /etc/modprobe.conf. Ubuntu has no doubt found a way to make that dark and muddy again so you might need to add it to an /etc/modprobe/whatever file instead. Please ask a fellow ubuntu victim for advice about translating this generic linux advice to your install. Once you do have that all setup, sound should "Just Work". You probably need to adjust the volume as specified above once only after which ubuntu will save it for the next boot, but if you don't hear anything ater rebooting, please check that first. Rene.

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/28/2007 07:06 AM, Cody Jung wrote: > As it suggests in the subject, I'm both a Linux-newbie and an > ALSA-newbie. I have an old clunker of a machine running Ubuntu Feisty > that I built out of parts left on curbs during spring cleanup, and that's > how I came across my sound card. It's an Ensoniq SoundscapeVivo (Not a > Vivo90, AFAIK). As far as I'm aware, there are no other VIVO's than the VIVO90. > I've gone through install and troubleshooting guides galore, but I can't > seem to get ALSA to recognize my card, and I'm confused and rather tired > of it all. > > Do you think you can help, and if so, what information do you need from > me? I probably can; I have both a Soundscape VIVO90 and a much older Soundscape S-2000. My VIVO90 is an ISA-PnP card and has a Ensoniq MARK5-RO, an OTTOR2C and a Analog Devices AD1845 codec: http://en.wikipedia.org/wiki/Ensoniq_Soundscape_VIVO90 I tested that when I acquired it a while ago and if I remember correctly, it just worked... mmm. come to think of it, it might be the case that I needed to add its PnP ID to the driver. Not sure anymore. Could you confirm whether or not your Soundscape is an ISA-PnP card and if so (that's also a way to find out if you don't know if it is) run "pnpdump" with it installed and look for a PnP-ID for the card in it? Something of the form "ENSxxxx". If your card is one of the older non-PnP soundscapes it's going to be a bit harder; they also need firmware and all that and I haven't tested that card myself yet. (please use the @alsa-project.org list addresses by the way; in the case of alsa-devel, the sourceforge one is gone even). Rene.

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[alsa-devel] Apple iMac pin configs
by Ivan N. Zlatev 28 Apr '07

28 Apr '07

Hello, I have a 20'' iMac machine with a hda-intel (SigmaTel 9221 A1 codec) and I have successfully been using the mac mini pin config, but there is one problem - the microphone doesn't work. The machine has a built-in microphone, built-in speakers, 1 digitial out jack and 1 for headphones. With the mac mini pin config I get 2 mics: "Front Mic" and "Mic" but none of those works. (alsamixer-macmini.gz for info). Today I spent some time on the search for the the correct pin configuration and it should be this [1] according to the windows drivers. This configuration causes amixer.gz, basically 1 "Mic" (which again doesn't record), but also "Mic as Output" and "Line in as output" (I don't have a line in as far as i know). I have also tried the macbook-pro and macbook introduced with 1.0.14rc3, but the mic doesn't work. I have attached * cat /proc/asound/card0/codec#0 * Mac OS X's complete dump of information of the audio driver - imac20-ioreg-audio.gz * amixer with the below config and the one with the macmini pin config Can you possible assist me to get the below pin config in tune, so that everything works? Regards. [1] ========== [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x0121E230 Action = hex: 01,0A,81,0C,81,10 [HKR\Settings\Pin\0B] CfgDflt = dword: 0x90A70120 [HKR\Settings\Pin\0C] CfgDflt = dword: 0x9017E110 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\0E] CfgDflt = dword: 0x400000FD Action = hex: 01,10,81,0A,81,0C [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0181E02E Action = hex: 09,0F [HKR\Settings\Pin\10] CfgDflt = dword: 0x1145E240 [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000FA [HKR\Settings\Pin\15] CfgDflt = dword: 0x400000FC [HKR\Settings\Pin\1B] CfgDflt = dword: 0x400000FB -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 27 Apr '07

27 Apr '07

Thanks to the help I've gotten on this list my alsa driver is working great. I got a little hung up not realizing that the alsa core basically provides everything you need to implement your own circular buffer if you need to use an intermediate buffer. The only question I still have is that if alsa allocates your DMA buffer for you based on snd_pcm_hardware_t buffer_bytes_max what exactly does snd_pcm_lib_preallocate_pages_for_all allocate buffers for? Does this allocate the buffers that would be passed to the copy callback? Thanks, Tim --- Tim Harvey <tim_harvey(a)yahoo.com> wrote: > Greetings, > > I'm setting out to write an ALSA driver for a very limited device thats > behind > a PLD. The actual codec is a uda1380 and does 16bit MSB stereo > encode/decode. > The way the device is driven the sampling rate is limited to 8KHz and data > transfer to/from the device I have to handle manually. > > I've read over the 'writing an ALSA driver' tutorial and have started the > driver but have run into a few questions: > > - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB > samples does the following snd_pcm_hardware_t look right?: > > static snd_pcm_hardware_t my_playback_hw = { > .info = (SNDRV_PCM_INFO_MMAP | > SNDRV_PCM_INFO_INTERLEAVED | > SNDRV_PCM_INFO_BLOCK_TRANSFER | > SNDRV_PCM_INFO_MMAP_VALID), > .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? > .rates = SNDRV_PCM_RATE_8000, // seems redundant? > .rate_min = 8000, > .rate_max = 8000, > .channels_min = 2, > .channels_max = 2, > .buffer_bytes_max = 32768, > .period_bytes_min = 4096, > .period_bytes_max = 32768, > .periods_min = 1, > .periods_max = 1024, > }; > > - I'm a little confused at how to deal with copying data from/to the device. > > It looks like I would implement the copy callback and copy the passed buffer > to > the device in the device specific manner I need to (playback), but I'm not > clear where to put data that I receive from the device (capture). I notice > that when the hw_params callback is called a dma buffer has been created by > ALSA in the substream->runtime struct. I assume this is created based on hte > buffer_bytes_max field above and I'm to copy data to this buffer? How do I > know where to copy it within the buffer and what data ALSA has already > consumed > from the buffer? > > - I might need to create an intermediate buffer between ALSA's rather large > buffers and my hardware buffer. Is there any driver that I could look at > that > does this? The tutorial talks about the vxpocket driver but that code looks > very foriegn from the rest of the ALSA drivers. > > - I'm not sure I understand what the 'period' is about. Its clear to me > that > I need to call snd_pcm_period_elapsed() periodically from either an IRQ or > other timed callback, but I'm not clear what the hardware pointer is or how > that converts to 'frames' and 'periods'. > > - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA > provides a mechanism for converting data from other formats? > > Thanks for any pointers, > > Tim >

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[alsa-devel] [ASoC] list corruption during AC97 registration
by Manuel Lauss 27 Apr '07

27 Apr '07

Hi Folks, I hit the following oops (on 2.6.21) when ASoC registers an AC97 codec. Happens both on MIPS and SH, for example the mips oops: asoc: AC97 HiFi <-> au1xpsc-ac97-1 mapping ok list_add corruption. prev->next should be next (80488f60), but was 865ffad8. (prev=865ffad8). <snip> Call Trace: [<80271290>] __list_add+0x68/0x94 [<802ac0e4>] bus_attach_device+0x48/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<c00882a8>] snd_soc_register_card+0x1d0/0x2c8 [snd_soc_core] [<c00b3268>] ac97_soc_probe+0x17c/0x1e4 [snd_soc_ac97] [<c0088f48>] soc_probe+0xb0/0x1ac [snd_soc_core] [<802ad410>] really_probe+0x148/0x268 [<802ac1cc>] bus_for_each_drv+0x50/0xa0 [<802ad758>] device_attach+0xe4/0x194 [<802ac0cc>] bus_attach_device+0x30/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<802aff24>] platform_device_add+0x13c/0x198 [<c00b71c8>] exm1200_mobo_ac97_init+0x1c8/0x1fc [snd_soc_exmmb_ac97] [<801478d0>] sys_init_module+0x1448/0x15b8 [<8010bcc0>] stack_done+0x20/0x3c System.map says: 80488e60 D ac97_bus_type As I said, it happens on both MIPS and SH (with machine and PCM drivers written by myself). With CONFIG_DEBUG_LIST disabled the code works and plays sound, but still I'd like to know what causes this list corruption. Thanks for any hints! -- Manuel Lauss

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[alsa-devel] dsnoop issues
by Gordon Miller 27 Apr '07

27 Apr '07

Hi All my snd_pcm_readi calls return too few frames. Also when I switch to stereo, the sound is muted. I cannot change this by adjusting period & buffer sizes. dsnoop is enabled by default. The documentation for snd_pcm_readi says that the only way that the frame count can be short is because of a signal or xrun. Well this clearly isn't the case. I think that direct snoop is the cause of this. Are there any remedies ? because I am roundly SICK of being told that my application must be wrong "because everyone else's works". I have spent the last 3 weeks debugging an app - that always worked properly - because of the faulty dsnoop implementation. It is abundantly clear that dsnoop is far from ready for use, but I am prepared to put some work into this. Whose idea was it to enable dsnoop by default, when it doesn't even work ? I was quite happy with OSS. Changing everything to be alsa entailed a lot of work, and I am beginning to think it was a complete waste of my time. ICE1712, 2.6.20.6 kernel, alsa 1.0.14rc1 Gordon

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[alsa-devel] New bugtracker categories
by Liam Girdwood 27 Apr '07

27 Apr '07

I'd like to request the following new categories for the ALSA bugtracker :- ASoC - platform - s3c24xx ASoC - platform - pxa2xx ASoC - platform - i.MX ASoC - platform - AT91xxxx ASoC - codecs ASoC - core Thanks Liam

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[alsa-devel] [ASoC] Sample format non available
by Markus Korber 27 Apr '07

27 Apr '07

Dear list, I'm currently implementing a ASoC solution for a custom chip and I've written a codec, I2S, PCM, and machine driver. However, when using aplay I get the following error (requested format was: 2): root:~# aplay -M -D hw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:906: Sample format non available However, I've told each piece involved, that it is capable of playing S16_LE. So I don't know what I am missing here? Should I provide more debugging output like RULES_DEBUG output from sound/core/pcm_native.c? ,----[ CODEC DAI ] | #define CS4265_RATES (SNDRV_PCM_RATE_8000_192000) | #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) | | struct snd_soc_codec_dai cs4265_dai = { | .name = "CS4265", | .playback = { | .stream_name = "Playback", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | .capture = { | .stream_name = "Capture", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | [...] | }; `---- ,----[ CPU DAI ] | #define CHIP_I2S_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ | SNDRV_PCM_RATE_48000) | #define CHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) | | struct snd_soc_cpu_dai chip_i2s_dai = { | /* DAI description */ | [...] | /* DAI callbacks */ | [...] | /* DAI capabilities */ | .playback = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | .capture = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | /* ops */ | [...] `---- ,----[ HW DAI ] | static const struct snd_pcm_hardware chip_pcm_hardware = { | .info = SNDRV_PCM_INFO_MMAP | | SNDRV_PCM_INFO_MMAP_VALID | /* For OSS emulation */ | SNDRV_PCM_INFO_BLOCK_TRANSFER | /* For OSS emulation */ | SNDRV_PCM_INFO_NONINTERLEAVED, | .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | | .channels_min = 2, /* Stereo */ | .channels_max = 2, | .period_bytes_min = 32, | .period_bytes_max = 8192, | .periods_min = 2, | .periods_max = 1024, /* PAGE_SIZE/sizeof(chip_dma_desc), */ | .buffer_bytes_max = 32 * 1024, | .fifo_size = 0, /* Still unused in ALSA? */ | }; `---- root:~# uname -a Linux 2.6.21 root:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). root:~# aplay --version aplay: version 1.0.14rc2 by Jaroslav Kysela <perex(a)suse.cz> root:~# aplay -M test.wav chip-pcm: Entered chip_pcm_open board_cs4265: Entered board_cs4265_startup Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_plug.c:840:(snd_pcm_plug_hw_refine_cchange) Unable to find an usable client format chip-pcm: Entered chip_pcm_hw_free chip-cs4265: Entered cs4265_shutdown chip-cs4265: Entered cs4265_write board_cs4265: Entered board_cs4265_shutdown chip-pcm: Entered chip_pcm_close ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IEC958_SUBFRAME_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: MU_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: A_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IMA_ADPCM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: GSM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: SPECIAL ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3BE aplay: set_params:878: Broken configuration for this PCM: no configurations available Regards, Markus Korber

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